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Taylor Brandstetter 3a034e15b4 Split DataChannel into two separate classes for RTP and SCTP.
Done in preparation for some threading changes that would be quite
messy if implemented with the class as-is.

This results in some code duplication, but is preferable to
one class having two completely different modes of operation.

RTP data channels are in the process of being removed anyway,
so the duplicated code won't last forever.

Bug: webrtc:9883
Change-Id: Idfd41a669b56a4bb4819572e4a264a4ffaaba9c0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178940
Commit-Queue: Taylor <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31691}
2020-07-10 00:03:21 +00:00
api Complete migration from "track" to "inbound-rtp" stats 2020-07-09 10:02:26 +00:00
audio webrtc::AudioSendStream: Add lock annotation to audio_level_ 2020-07-06 17:05:25 +00:00
build_overrides Rename sanitizers suppression files. 2020-07-03 18:29:58 +00:00
call [Adaptation] Move AdaptationConstraints to VideoStreamAdapter 2020-07-09 13:06:56 +00:00
common_audio Replace slave -> helper, master -> reference 2020-06-29 12:18:05 +00:00
common_video Migrate common_video/ and examples/ to webrtc::Mutex. 2020-07-07 13:33:27 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
docs Search and replace gendered terms according to style guide: 2020-06-12 14:12:54 +00:00
examples Migrate common_video/ and examples/ to webrtc::Mutex. 2020-07-07 13:33:27 +00:00
logging Search and replace gendered terms according to style guide: 2020-06-12 14:12:54 +00:00
media Migrate media/ to webrtc::Mutex. 2020-07-07 13:46:47 +00:00
modules Add skeleton of new capturer that uses Windows.Graphics.Capture API 2020-07-09 17:49:11 +00:00
p2p Revert "Implement packets_(sent | received) for RTCTransportStats" 2020-07-08 09:42:41 +00:00
pc Split DataChannel into two separate classes for RTP and SCTP. 2020-07-10 00:03:21 +00:00
resources iSAC API wrapper unit test fix 2020-02-27 14:27:23 +00:00
rtc_base Migrate rtc_base to webrtc::Mutex. 2020-07-08 20:38:54 +00:00
rtc_tools Roll chromium_revision 4d95e6c77b..71a0e1904e (776481:782339) 2020-06-26 05:33:14 +00:00
sdk Migrate rtc_base to webrtc::Mutex. 2020-07-08 20:38:54 +00:00
stats Complete migration from "track" to "inbound-rtp" stats 2020-07-09 10:02:26 +00:00
style-guide Add style guide rule about paired .h and .cc files 2018-03-14 13:02:35 +00:00
system_wrappers Use absl_deps in order to preapre to the Abseil component build release. 2020-06-08 12:59:40 +00:00
test Migrate a leftover in test/ to webrtc::Mutex. 2020-07-09 14:28:21 +00:00
tools_webrtc Fix 'Perf Android32 (M Nexus5)' config in MB. 2020-07-08 15:04:10 +00:00
video [Adaptation] Move AdaptationConstraints to VideoStreamAdapter 2020-07-09 13:06:56 +00:00
.clang-format Tune ObjC clang-format configuration 2017-05-11 09:14:18 +00:00
.git-blame-ignore-revs Let git-hyper-blame ignore new format cleanup. 2019-07-11 16:18:51 +00:00
.gitignore Add .clangd to .gitignore 2019-10-28 12:27:50 +00:00
.gn Reenable libaom decoder by default 2020-03-18 18:04:41 +00:00
.vpython Add source-side perf upload script for WebRTC. 2019-11-18 14:37:01 +00:00
abseil-in-webrtc.md Use absl_deps in order to preapre to the Abseil component build release. 2020-06-08 12:59:40 +00:00
AUTHORS Relanding: Fix data channel message integrity violation 2020-07-07 03:06:24 +00:00
BUILD.gn Convert GN libs lists to frameworks 2020-07-06 10:08:09 +00:00
CODE_OF_CONDUCT.md Add code of conduct to WebRTC repo 2017-05-16 12:09:13 +00:00
codereview.settings Don't add webrtc-reviews@ to CC, it can be added globally on Gerrit 2018-10-25 08:19:53 +00:00
common_types.h Replaces OverheadObserver with simple getter. 2020-05-07 17:33:45 +00:00
DEPS Roll chromium_revision b5359525fc..a69b9c614c (785857:786202) 2020-07-08 10:14:08 +00:00
ENG_REVIEW_OWNERS Enforce LGTM from owners of depends-on paths in DEPS via presubmit. 2018-09-28 12:49:54 +00:00
LICENSE Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
license_template.txt Update template to follow chromium copyright style 2013-04-24 01:01:28 +00:00
native-api.md Make transient suppression optionally excludable via defines 2020-04-02 11:44:07 +00:00
OWNERS Remove phoglund as root owner. 2020-03-30 12:15:56 +00:00
PATENTS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
PRESUBMIT.py Inclusive language in PRESUBMIT.py. 2020-06-30 14:03:10 +00:00
presubmit_test.py Use source_sets in component builds and static_library in release builds. 2019-10-17 21:17:18 +00:00
presubmit_test_mocks.py Reland: Add presubmit check for changes in 3pp 2018-05-22 13:11:18 +00:00
pylintrc Fixing py lint errors 2018-07-23 15:28:48 +00:00
README.chromium Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
README.md Fix link in documentation. (take 2) 2020-04-16 11:08:43 +00:00
style-guide.md C++ style: We don't allow designated initializers 2020-06-03 09:11:09 +00:00
WATCHLISTS Remove benwright@webrtc.org from WATCHLISTS 2020-01-31 18:46:52 +00:00
webrtc.gni Revert "Build: Disable the iLBC audio codec by default" 2020-07-09 11:27:06 +00:00
webrtc_lib_link_test.cc Rewrite the lib link test to just be a binary. 2019-10-18 07:42:20 +00:00
whitespace.txt Trigger bots. 2020-06-15 08:40:44 +00:00

WebRTC is a free, open software project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. The WebRTC components have been optimized to best serve this purpose.

Our mission: To enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols.

The WebRTC initiative is a project supported by Google, Mozilla and Opera, amongst others.

Development

See here for instructions on how to get started developing with the native code.

Authoritative list of directories that contain the native API header files.

More info